streaming delay

Tim Müller tim at centricular.com
Wed Feb 12 10:33:20 PST 2014


On Wed, 2014-02-12 at 16:33 +0100, Lee Matthews wrote:

> I'm guessing it's because flacenc needs a certain number of samples
> before it can compress a block, it therefore has to wait longer with a
> lower sample rate ? 
> 
> Is there any way that I can reduce the sample rate whilst keeping a low delay ?

You can play with flacenc's "blocksize" property, see gst-inspect-1.0
flacenc.

Have you measured the latency without any encoder at all? That is just
plain audio/x-raw over TCP ? You can parse it on the other end with
audioparse, see gst-inspect-1.0 audioparse.

For low-latency streaming maybe also consider UDP/RTP (there's no flac
payloader but there's a generic gst payloader which should work in that
case, and if it doesn't have to be lossless, maybe consider Opus, as
pointed out already by someone).

Cheers
 -Tim

-- 
Tim Müller, Centricular Ltd - http://www.centricular.com



More information about the gstreamer-devel mailing list